The proliferation of data transport networks, most notably the Internet, is causing a revolution in telephony and other forms of real-time communication. Businesses that have been accustomed to having telephony traffic and data traffic separately supported over different systems and networks are now moving towards so-called “converged networks” wherein telephone voice traffic and other forms of real-time media are converted into digital form and carried by a packet data network along with other forms of data. Now that the technologies are feasible to support it, voice over data transport offers many advantages in terms of reduced capital and operating costs, resource efficiency and flexibility.
For example, at commercial installations, customer premise equipment investments are substantially reduced as most of the enhanced functions, such as PBX and automatic call distribution functions, may reside in a service provider's network. Various types of gateways allow for sessions to be established even among diverse systems such as IP phones, conventional analog phones and PBXs as well as with networked desktop computers.
To meet the demand for voice over data transport, service providers and network equipment vendors are faced with the challenges of establishing new protocols and standards, recognizing new business models, implementing new services, and designing new equipment in a way that would have been difficult to imagine twenty years ago.
For example, a new generation of end user terminal devices are now replacing the traditional telephones and even the more recent PBX phone sets. These new sets, such as those offered by Cisco and Pingtel, may connect directly to a common packet data network, via an Ethernet connection for example, and feature large visual displays to enhance the richness of the user interface.
Even before such devices were developed, computers equipped with audio adapters and connected to the Internet were able to conduct some rudimentary form of Internet telephony, although the quality was unpredictable and often very poor. The emphasis now is upon adapting internet protocol (IP) networks and other packet transport networks to provide reliable toll-quality connections, easy call set-up and enhanced features to supply full-featured telephony as well as other forms of media transport. Some other types of media sessions enabled by such techniques may include video, high quality audio, multi-party conferencing, messaging and collaborative applications.
Of course, as a business or residential communications subscriber begins using such voice-over-packet communications to replace conventional telephony, there will naturally be an expectation that the quality of the connections and the variety of services will be at least as good as in the former telephone network. In terms of services, for example, some businesses have come to rely upon PBX features or network-resident “Centrex” features such as call forwarding and conditional call handling. In the near future, such special services are expected to see increased use because the new terminal devices mentioned earlier can provide a much more intuitive interface for the users. With existing systems, users often forget which combinations of keystrokes are required to invoke enhanced features.
For establishing a communications session in a network, new protocols and control architectures have emerged. It is worth noting that these have been inspired by the migration to a voice over data but are not necessarily limited to such an environment. In some respects the protocols and control architectures described next may be used to establish calls through any form of transport.
Both the ITU H.323 standard and the IETF's Session Initiation Protocol (SIP) are examples of protocols which may be used for establishing a communications session among terminals connected to a network. The SIP protocol is described in IETF document RFC 2543 and its successors. Various architectures have been proposed in conjunction with these protocols with a common theme of having an address resolution function, referred to as a “location server,” somewhere in the network to maintain current information on how to reach any destination and to control features on behalf of users.
For large scale-deployment of voice over data transport as well as other real-time communications, it is essential that the network control architectures be extremely robust and highly scalable to reliably accommodate millions of sessions on a daily basis. Robustness may necessitate designing in redundancy and failover mechanisms. Preferably, these measures will even provide transparent continuity of existing sessions and features even if a failure occurs in the midst of a session. For ensuring this level of reliability and for maximizing scalability, it is generally preferable to minimize the demand upon control functions, such as location servers, to maintain any persistent state information for each call in the network.